similar to: How to integrate Asterisk with XMPP

Hi, Do you really think Presence should be used to forward call to voicemail ? My feeling is forwarding incoming calls to voicemail should remain a different task as you could wish to remain unavailable for chat and still reachable by phone. As I can't see a way to define Presence status such as "unavailable for chat and phones", "unavailable for chat but available for

We have used AIS for disturbed Device State in the past, BLF and MWI, We are in the process of an update on one of our clustered systems, We are looking at XMPP and I found a few discussions on a Corosync with has OpenAIS built in. My question is which should I be looking at to replace my current AIS option I currently have. XMPP or Corosync? It looks like the Corosync is just the

I just get started with it so my question maybe not well catch. Anyway to do the VOIP call and IM we need to use two difference servers? which one is asterisk for VOIP ? and other one for IM that is openfire ? or we can have other choice better than this ? Thank you for your help, I am waiting for your reply. Thyda On Wed, Jul 8, 2015 at 12:43 PM, Kristof Van Den Ouweland < kvandenouweland

Actually, I am using the openfire and I create two users with the SIP mapping on the openfire to the asterisk server. I can register one user with the openfire client(Spark) and yes it is connect to asterisk SIP also. But with the other one user, I register it with the SIP client(Zoiper/ or Linphone) and then I can make the call over these two SIP but they cannot reach the chat. I wonder what

I have been working with distributed device states in Asterisk using XMPP attached to an OpenFire server. I have it working well across two servers and want to roll it out across every server in my company. All servers are Asterisk 11.6.0. I am running into a problem that seems like it should be a bit easier to solve than it is seeming to be. On the third server I am rolling into this

The Asterisk Development Team has announced the release of Asterisk 1.8.2. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.2 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this release:

The Asterisk Development Team has announced the release of Asterisk 1.8.2. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.2 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this release:

Given the recent announcement about Google slimming their support for public interconnection with XMPP, can anybody comment on where this leaves the XMPP support in Asterisk? In particular, I notice many of the references to XMPP on the wiki link to https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google which seems to suggest that XMPP support and Google Talk support are one and the

Hi all, Do you have an idea on the best way on how to implement a system with multiple Asterisk servers with BLF working in such a way that a peer on one server can subscribe to another peer on the other server in a seamless manner? Has anyone set-up a system like this before? Thanks! Regards, Ronald -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hmm, most phones I've used wouldn't have the capability of displaying a bitmap image due to only having minimal monochrome displays. What sort of end device do you perceive to display these images? Can you give links to any devices with support for such things? I'm assuming you mean only a still-photo image, not video image. Perhaps you could use a video channel for this and simply

Hi list, I have a big doubt!, I have some users with ejabberd and am using motif to make some calls to extensions, here works fine, the problem is when I want to send a message to another user on ejabberd and asterisk take this message as part him, like a sip message , the other user does not receive this message xmpp User A xmpp == Chat to == User B xmpp (not receive the message) look cli

Hi all, For one of my inverstigations it looks like i'm back to "square one" I'm trying to accept an incoming xmpp call and forward it conditionally to a sip, isdn, or voicemail. No google is involved as i use a local xmpp server (ejabberd) I was experimenting on 1.8.15.1 (with jabber.conf, jingle.conf), but some suggested me to have a look at asterisk11,so i did... I

I've been working on an interactive XMPP interface so users at my office can interact with the timeclock and queues by XMPP (in addition to IVR menu, which has been running just fine for quite a while before the XMPP interface). I'm using sendtodialplan=yes to handling the incoming unsolicited messages, and typically will have at least one point of interaction where Asterisk requests

I'm trying to set up a way that our users can send an XMPP message to Asterisk (unsolicited) to request information, such as voicemail status or the like. No matter what I set for the dialplan, I'm only seeing Asterisk execute the s,1 priority in the context defined in xmpp.conf for incoming messages, and then the "call" hangs up without executing further instructions. Anything

Hello all, What is the best way, using sieve, to send XMPP notifications? I am using Debian Stretch, with Dovecot core / sieve packages version 2.2.27-3+deb9u2. Should I use the enotify extension, or a script with extprograms extension? Thanks for your answers. -- Andr? Rodier

Hello, I'm looking for a SIP to XMPP Jingle voice gateway. I see that Asterisk has Jabber and Jingle support, but it looks like Asterisk acts as a Jabber client. Are there any Jabber server solutions, where Jabber users can call SIP users by using the SIP URI and vice versa? -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.com/

Friends, a new dialplan application is now available for testing : http://svn.digium.com/view/asterisk/team/phsultan/jabberreceive/ The corresponding feature request is located here : http://bugs.digium.com/view.php?id=12569 What can you do with it? Well, a direct usage of this application is to make an easy to use GoogleTalk voice gateway out of Asterisk. Here is an example (assuming the

I've written a new article about Asterisk 1.4's Jabber integration. Check it out at http://www.voip-forum.com/asterisk/2008-01/xmpp/ /Olle

Hello, (shamelessly adapted to my needs from a old message from this list) I want to integrate status messages over xmpp in my Rails application. My problem is that I do not want to establish a connection to the jabber server each time a message is sent and disconnect the connection afterwards. So I am looking for a solution to establish the connection and use this connection in my controllers

I've set up google voice to chat with me: Forwards calls to: <me>@gmail.com and xmpp: [general] debug=no ; Enable debugging (disabled by default). autoprune=yes ; Auto remove users from buddy list. Depending on your ; setup (ie, using your personal Gtalk account for a test)